Traditionally mobile network base transceiver stations (BTS) have exchanged data with the core mobile network via a dedicated, high capacity connection to an associated base station controller (BSC), e.g., a dedicated T-1/E-1 line. In some cases, it may be desirable to use an IP or other packet data network to enable a BTS to exchange data with a BSC. However, to meet quality of service obligations to carriers and/or provide a satisfactory call experience to users, care must be taken to ensure call data is communicated in an efficient manner that ensures safe and timely receipt at the destination.
One challenge faced when transmitting call data between a base transceiver station and a base station controller via a packet data network is that transmission times across such networks may vary over the short term, e.g., due to variations in the volume of network traffic being sent at a particular time; changing environmental, workload, or other conditions affecting one or more nodes in the network path; singular and/or periodic events that affect the availability and/or speed of one or more nodes; etc. This characteristic of packet data networks, known as “jitter”, makes it difficult or often impossible to predict with certainty the time it will take for a given packet sent by a sending node to reach its destination. However, typically a mobile telecommunication protocol requires that a packet be transmitted at a prescribed interval (e.g., one every 20 msec in the case of GSM), and it would not be desirable to propagate network jitter to call data destinations, which could result in audible manifestations perceived by a user, such as by garbling or “breaking up” call voice data. Therefore, there is a need for a way to manage the effect of jitter on a packet data network used to transport mobile network data.